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Update NetEq bitexactness tests to only run on Linux.

Browse Source 
Running bitexactness tests only on Linux makes it significantly easier to
update them, while still giving many of the same benefits.

Bug: webrtc:12518, b/216736217
Change-Id: I7f3c9a27c0fc14b7ee0e83aede2e7702cfa79141
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/249787
Reviewed-by: Jakob Ivarsson‎ <jakobi@webrtc.org>
Commit-Queue: Ivo Creusen <ivoc@webrtc.org>
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35829}
This commit is contained in:
Ivo Creusen
2022-01-28 17:07:09 +01:00
committed by WebRTC LUCI CQ
parent 5b66130209
commit a37384899b
2 changed files with 85 additions and 312 deletions
Download Patch File Download Diff File Expand all files Collapse all files

86
modules/audio_coding/neteq/neteq_unittest.cc
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@ -44,36 +44,10 @@ ABSL_FLAG(bool, gen_ref, false, "Generate reference files.");
namespace webrtc {
namespace {
const std::string& PlatformChecksum(const std::string& checksum_general,
const std::string& checksum_android_32,
const std::string& checksum_android_64,
const std::string& checksum_win_32,
const std::string& checksum_win_64) {
#if defined(WEBRTC_ANDROID)
#ifdef WEBRTC_ARCH_64_BITS
return checksum_android_64;
#else
return checksum_android_32;
#endif // WEBRTC_ARCH_64_BITS
#elif defined(WEBRTC_WIN)
#ifdef WEBRTC_ARCH_64_BITS
return checksum_win_64;
#else
return checksum_win_32;
#endif // WEBRTC_ARCH_64_BITS
#else
return checksum_general;
#endif // WEBRTC_WIN
}
} // namespace
#if !defined(WEBRTC_IOS) && defined(WEBRTC_NETEQ_UNITTEST_BITEXACT) && \
(defined(WEBRTC_CODEC_ISAC) || defined(WEBRTC_CODEC_ISACFX)) && \
defined(WEBRTC_CODEC_ILBC) && !defined(WEBRTC_ARCH_ARM64)
#if defined(WEBRTC_LINUX) && defined(WEBRTC_ARCH_X86_64) && \
defined(WEBRTC_NETEQ_UNITTEST_BITEXACT) && \
(defined(WEBRTC_CODEC_ISAC) || defined(WEBRTC_CODEC_ISACFX)) && \
defined(WEBRTC_CODEC_ILBC)
#define MAYBE_TestBitExactness TestBitExactness
#else
#define MAYBE_TestBitExactness DISABLED_TestBitExactness
@ -83,66 +57,50 @@ TEST_F(NetEqDecodingTest, MAYBE_TestBitExactness) {
webrtc::test::ResourcePath("audio_coding/neteq_universal_new", "rtp");
const std::string output_checksum =
PlatformChecksum("ba4fae83a52f5e9d95b0910f05d540114285697b",
"aa557f30f7fdcebbbbf99d7f235ccba3a1c98983", "not used",
"ba4fae83a52f5e9d95b0910f05d540114285697b",
"64b46bb3c1165537a880ae8404afce2efba456c0");
"ba4fae83a52f5e9d95b0910f05d540114285697b";
const std::string network_stats_checksum =
PlatformChecksum("fa878a8464ef1cb3d01503b7f927c3e2ce6f02c4",
"300ccc2aaee7ed1971afb2f9a20247ed8760441d", "not used",
"fa878a8464ef1cb3d01503b7f927c3e2ce6f02c4",
"fa878a8464ef1cb3d01503b7f927c3e2ce6f02c4");
"fa878a8464ef1cb3d01503b7f927c3e2ce6f02c4";
DecodeAndCompare(input_rtp_file, output_checksum, network_stats_checksum,
absl::GetFlag(FLAGS_gen_ref));
}
#if !defined(WEBRTC_IOS) && defined(WEBRTC_NETEQ_UNITTEST_BITEXACT) && \
defined(WEBRTC_CODEC_OPUS)
#if defined(WEBRTC_LINUX) && defined(WEBRTC_ARCH_X86_64) && \
defined(WEBRTC_NETEQ_UNITTEST_BITEXACT) && defined(WEBRTC_CODEC_OPUS)
#define MAYBE_TestOpusBitExactness TestOpusBitExactness
#else
#define MAYBE_TestOpusBitExactness DISABLED_TestOpusBitExactness
#endif
// TODO(http://bugs.webrtc.org/12518): Enable the test after Opus has been
// updated.
TEST_F(NetEqDecodingTest, DISABLED_TestOpusBitExactness) {
TEST_F(NetEqDecodingTest, MAYBE_TestOpusBitExactness) {
const std::string input_rtp_file =
webrtc::test::ResourcePath("audio_coding/neteq_opus", "rtp");
const std::string maybe_sse =
"c7887ff60eecf460332c6c7a28c81561f9e8a40f"
"|673dd422cfc174152536d3b13af64f9722520ab5";
const std::string output_checksum = PlatformChecksum(
maybe_sse, "e39283dd61a89cead3786ef8642d2637cc447296",
"53d8073eb848b70974cba9e26424f4946508fd19", maybe_sse, maybe_sse);
const std::string output_checksum =
"6e23d8827ae54ca352e1448ae363bdfd2878c78e";
const std::string network_stats_checksum =
PlatformChecksum("c438bfa3b018f77691279eb9c63730569f54585c",
"8a474ed0992591e0c84f593824bb05979c3de157",
"9a05378dbf7e6edd56cdeb8ec45bcd6d8589623c",
"c438bfa3b018f77691279eb9c63730569f54585c",
"c438bfa3b018f77691279eb9c63730569f54585c");
"f89a9533dbb35a4c449b44c3ed120f7f1c7f90b6";
DecodeAndCompare(input_rtp_file, output_checksum, network_stats_checksum,
absl::GetFlag(FLAGS_gen_ref));
}
// TODO(http://bugs.webrtc.org/12518): Enable the test after Opus has been
// updated.
TEST_F(NetEqDecodingTest, DISABLED_TestOpusDtxBitExactness) {
#if defined(WEBRTC_LINUX) && defined(WEBRTC_ARCH_X86_64) && \
defined(WEBRTC_NETEQ_UNITTEST_BITEXACT) && defined(WEBRTC_CODEC_OPUS)
#define MAYBE_TestOpusDtxBitExactness TestOpusDtxBitExactness
#else
#define MAYBE_TestOpusDtxBitExactness DISABLED_TestOpusDtxBitExactness
#endif
TEST_F(NetEqDecodingTest, MAYBE_TestOpusDtxBitExactness) {
const std::string input_rtp_file =
webrtc::test::ResourcePath("audio_coding/neteq_opus_dtx", "rtp");
const std::string maybe_sse =
"0fb0a3d6b3758ca6e108368bb777cd38d0a865af"
"|79cfb99a21338ba977eb0e15eb8464e2db9436f8";
const std::string output_checksum = PlatformChecksum(
maybe_sse, "b6632690f8d7c2340c838df2821fc014f1cc8360",
"f890b9eb9bc5ab8313489230726b297f6a0825af", maybe_sse, maybe_sse);
const std::string output_checksum =
"5cea4a8e750842ac67b79e8e2ce6a0a1c01f8130";
const std::string network_stats_checksum =
"18983bb67a57628c604dbdefa99574c6e0c5bb48";
"dc8447b9fee1a21fd5d1f4045d62b982a3fb0215";
DecodeAndCompare(input_rtp_file, output_checksum, network_stats_checksum,
absl::GetFlag(FLAGS_gen_ref));